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It means that some peaks in your audio are so high, the average loudness can not be raised till -23 LUFS. You should manually or with the aid of a limiter lower the highest peaks and run LevelNorm again. Mark that these peaks are often very short, they usually do not have any meaning for the perception of your program so you can safely remove them.
The target level for EBU R128 is -23 LUFS, which is quite low in comparison to old style ‘cranked up’ broadcast mixes. There is no problem with that, during broadcast your mix will sound equally loud as other mixes. If you like to take advantage of the new standard, apply less limiting and compression and use a real time meter like LevelView to help you getting the right balance. As a rule of thumb: peek at the large integrated loudness number once in a while to keep it around ‘-23 LUFS’ and make sure that normal dialog drives the rainbow meter around 0 LU. Don’t be afraid if the meter is temporarily high or low on your full mix. When you’re done, correct the master volume of the mix to make the meter match -23 LUFS exactly or use LevelNorm to do that automatically.
We advice to use the AAX version in PT 10 to ensure the correct initialisation of the plugin for each file that is measured. The RTAS will retire soon.
Plugins on windows will be installed in the default locations:
- RTAS for windows 32 bit:
- C:\Program Files\Common Files\Digidesign\DAE\Plug-Ins\
- RTAS for windows 64 bit:
- C:\Program Files (x86)\Common Files\Digidesign\DAE\Plug-Ins\
- AAX for windows 32 bit:
- C:\Program Files (x86)\Common Files\Avid\Audio\Plug-Ins\
- AAX for windows 64 bit:
- C:\Program Files\Common Files\Avid\Audio\Plug-Ins\
If your mix mainly consists of dialog, a less precise but much faster method is to normalise the main voice channel and mix the other channels in balance around this. For ATSC compliance this is actually the preferred way of working. In that case, follow the instructions above, but select the voice track, and make sure LevelNorm is set to ‘mono’ format if your track is mono. This method also works in rushed environments like news editing where often no time is left to normalize the stereo end mix. It also helps to rapidly match the loudness of various tracks like interviews from different origins. Do this by selecting all voice files on the same track and apply LevelNorm to them. They will be analysed and rendered independently. Mark that Avid editor software demands that a new render of AudioSuite plugins is performed every time the audio clips have been editted. To avoid repeated rendering, an alternative working method is to use only the “Find LU” function of LevelNorm and apply the necessary gain change to reach 0 LU manually with the Avid clip gain function. This gain change will not be affected by editing. Mind to keep the fader of the dialog channel to 0 dB.
Normally your target level will be 0 LU. When using LevelNorm in the course of a mix process however, it can be convenient to normalize for instance background music to your favorite level below 0 LU. This may speed up your production time considerably. To use this function, just select the preferred target level with the slider before clicking ‘find LU’ or ‘Render’.
LevelNorm is an AudioSuite plugin and therefore operates on files, not in real time. This file needs to actually exist in the bin (or timeline) and it must be a stereo file if the broadcast is in stereo. In an Avid video editor environment, AudioSuite can only process one file at a time and therefore two mono files do not make a stereo pair. Fortunately, starting from Avids Media Composer 5 stereo tracks can be used in the timeline. Note however that the default track type is still mono. Consult the Media Composer manual on how to create stereo tracks. To learn more about the stereo feature you may also like to view this online tutorial here (3rd party).
LevelNorm automatically tests whether your audio is mono or stereo. In Avid video workstations the surround formats are of no use, so you can ignore the ‘format’ option.
To normalize your audio, the first step is to print your mix to a new stereo file. This is most easily done using the menu “Special -> Audio Mixdown” and then select “Stereo” and a new audio track. Now open the AudioSuite toolbox from the ‘tools’ menu. Select the stereo audio file on the time line or drop it from the bin on the AudioSuite toolbox. Select LevelNorm-EBU or LevelNorm-ATSC from the plugin menu within the AudioSuite toolbox. Open and activate the plugin interface by clicking the purple plugin icon. The user interface looks like the screenshot above. Select your desired format and target level in the LevelNorm interface (usually ‘stereo’ and ‘0 LU’). Next click ‘Render’ and wait for the analysis and rendering to finish. Click OK. The file is now leveled to the selected target. If the average loudness of the file was so low that normalizing it to 0 LU would cause a clip, LevelNorm will show a warning and amplify the file to a maximum peak level of 0 dBTP. Mark that to maintain the correct 0 LU level of a normalized track during a second print, the channel fader should be set to 0 dB. LevelNorm also offers an option to just analyse a selected audio file’s loudness. To do so, click ‘Find LU’ in stead of ‘Render’. After analysis, the plugin window shows the file’s loudness but has not changed its level.
We advice to use the AAX version in PT 10 to ensure the correct initialisation of the plugin for each file that is measured.
LevelNorm is an AudioSuite plugin and therefore only operates on a selected file or part of a file and not in real time on a channel of your virtual mixing desk. Consequence is it will ignore the fader positions of your mixing desk. This means that if you like to use LevelNorm for making your final mix comply to the EBU or ATSC recommendation, first you need to print your mix to a separate mono, stereo or surround track. Operation then is as usual with Audio Suite plugins: select the audio file and open LevelNorm from the AudioSuite menu (it can be found in the ‘Grimm Audio’ section). The LevelNorm User Interface window now opens, as seen below.
LevelNorm automatically sets the amount of channels to mono, stereo or surround, dependent on the source. In case your source is surround, please make sure that the ‘format’ is set to the right channel order. You can change this setting by clicking the window next to the text ‘format’. A drop down menu will open with a choice between SMPTE, Film and DTS channel order.
AudioSuite processing in Protools has relatively large freedom in selecting files and part of files. For instance for stereo either a true stereo track or two mono tracks can be selected. It is also possible to select part of a file or even more than one file on the timeline and analyse or process the total amount of selected audio.
After the track count format, set the target level. With LevelNorm-EBU, 0 LU equals -23 LUFS. For LevelNorm-ATSC, 0 LU is -24 LKFS. Normally your target will be 0 LU. When using LevelNorm in the course of your mix process however, it can be convenient to normalize for instance background music to your favorite level below 0 LU. This may speed up your production time considerably. To use this function, just select the preferred target level with the slider before clicking ‘Process’.
Next step is to either only ‘Find LU’ of the file, or directly ‘Process’ it to the loudness target level set with the slider. A progress bar will be shown during the calculation and the normalized file will replace the original file on the timeline. If the average loudness of the file was so low that normalizing it to 0 LU would cause a clip, LevelNorm will show a warning and amplify the file to a maximum peak level of 0 dBTP. Mark that to maintain the correct 0 LU level of a normalized track during a second print, the channel fader and master fader should be set to 0 dB.
Normally your target level will be 0 LU. When using LevelNorm in the course of a mix process however, it can be convenient to normalize for instance background music to your favorite level below 0 LU. This may speed up your production time considerably. To use this function, just select the preferred target level with the slider before clicking ‘Process’.
The AAX version of LevelNorm-ATSC has an extra option: “Dialog Gate”. This enables ‘Dialog Intelligence’ (by Dolby), a process that automatically detects speech and only enables the loudness measurement for audio fragments in which speech is present. Automatic speech detection is a difficult task and therefore an algorithm like this can not replace manual detection perfectly. But if your aim is to use speech as an anchor element, you may use LevelNorm-ATSC for this.
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